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Page 196 / 275 Scroll up to view Page 191 - 195
Vigor2110 Series User’s Guide
189
這邊的內容與
user mode
底下不同，這是對的嗎
?
SIP Port
Set the port number for sending/receiving SIP message for
building a session. The default value is
5060.
Your peer must
set the same value in his/her Registrar.
Domain/Realm
Set the domain name or IP address of the SIP Registrar server.
Proxy
Set domain name or IP address of SIP proxy server. By the time
you can type
:port
number
after the domain name to specify
that port as the destination of data transmission (e.g.,
nat.draytel.org:5065
)
Act as Outbound Proxy
Check this box to make the proxy acting as outbound proxy.
Display Name
The caller-ID that you want to be displayed on your friend’s
screen.
Account Number/Name
Enter your account name of SIP Address, e.g. every text before
@.
Authentication ID
Check the box to invoke this function and enter the name or
number used for SIP Authorization with SIP Registrar. If this
setting value is the same as Account Name, it is not necessary
for you to check the box and set any value in this field.
Password
The password provided to you when you registered with a SIP
service.
Expiry Time
The time duration that your SIP Registrar server keeps your
registration record. Before the time expires, the router will send
another register request to SIP Registrar again.
NAT Traversal Support
If the router (e.g.
,
broadband router) you use connects to
internet by other device, you have to set this function for your
necessity.
None –
Disable this function.
Stun
– Choose this option if there is Stun server provided for
your router.
Manual
– Choose this option if you want to specify an external
IP address as the NAT transversal support.
Nortel
– If the soft-switch that you use supports Nortel solution,
you can choose this option.
Ring Port
Set Phone1 or Phone2 as the default ring port for this SIP
account.
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Vigor2110 Series User’s Guide
190
Ring Pattern
Choose a ring tone type for the VoIP phone call.
4.12.3 Phone Settings
This page allows user to set phone settings for Phone 1 and Phone 2 respectively. However, it
changes slightly according to different model you have.
Phone List
Port – t
here are two phone ports provided here for you to
configure.
Phone1/Phone2
allow you to set general settings for
PSTN phones. Please refer to
Section 4-1
for detailed
information of ISDN phone/network connection.
Call Feature
– A brief description for call feature will be
shown in this field for your reference.
Codec
– The default Codec setting for each port will be shown
in this field for your reference. You can click the number below
the Index field to change it for each phone port.
Tone
- Display the tone settings that configured in the advanced
settings page of Phone Index.
Gain
- Display the volume gain settings for Mic/Speaker that
configured in the advanced settings page of Phone Index.
Default SIP Account
– “draytel_1” is the default SIP account.
You can click the number below the Index field to change SIP
account for each phone port.
DTMF Relay
– Display DTMF mode that configured in the
advanced settings page of Phone Index.
RTP
Symmetric RTP
– Check this box to invoke the function. To
make the data transmission going through on both ends of local
router and remote router not misleading due to IP lost (for
Page 198 / 275
Vigor2110 Series User’s Guide
191
example, sending data from the public IP of remote router to the
private IP of local router), you can check this box to solve this
problem.
Dynamic RTP Port Start
- Specifies the start port for RTP
stream. The default value is 10050.
Dynamic RTP Port End
- Specifies the end port for RTP
stream. The default value is 15000.
RTP TOS
– It decides the level of VoIP package. Use the drop
down list to choose any one of them.
Detailed Settings for Phone Port
Click the number link for Phone port, you can access into the following page for configuring
Phone settings.
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Vigor2110 Series User’s Guide
192
Hotline
Check the box to enable it. Type in the SIP URL in the field for
dialing automatically when you pick up the phone set.
Session Timer
Check the box to enable the function. In the limited time that
you set in this field, if there is no response, the connecting call
will be closed automatically.
Call Forwarding
There are four options for you to choose.
Disable
is to close call
forwarding function.
Always
means all the incoming calls will
be forwarded into SIP URL without any reason.
Busy
means
the incoming calls will be forwarded into SIP URL only when
the local system is busy.
No Answer
means if the incoming
calls do not receive any response, they will be forwarded to the
SIP URL by the time out.
SIP URL
– Type in the SIP URL (e.g., [email protected] or
[email protected]) as the site for call forwarded.
Time Out
– Set the time out for the call forwarding. The
default setting is 30 sec.
DND (Do Not Disturb)
mode
Set a period of peace time without disturbing by VoIP phone
call. During the period, the one who dial in will listen busy
tone, yet the local user will not listen any ring tone.
Index (1-15) in Schedule -
Enter the index of schedule profiles
to control the DND mode according to the preconfigured
schedules. Refer to section
3.8.2 Schedule
for detailed
configuration.
Index (1-60) in Phone Book -
Enter the index of phone book
profiles. Refer to section
3.11.1 DialPlan – Phone Book
for
detailed configuration.
CLIR (hide caller ID)
Check this box to hide the caller ID on the display panel of the
phone set.
Call Waiting
Check this box to invoke this function. A notice sound will
appear to tell the user new phone call is waiting for your
response. Click hook flash to pick up the waiting phone call.
Call Transfer
Check this box to invoke this function. Click hook flash to
initiate another phone call. When the phone call connection
succeeds, hang up the phone. The other two sides can
communicate, then.
Prefer Codec
Select one of five codecs as the default for your VoIP calls. The
codec used for each call will be negotiated with the peer party
before each session, and so may not be your default choice. The
default codec is G.729A/B; it occupies little bandwidth while
maintaining good voice quality.
If your upstream speed is only 64Kbps, do not use G.711 codec.
It is better for you to have at least 256Kbps upstream if you
would like to use G.711.
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Vigor2110 Series User’s Guide
193
Single Codec
– If the box is checked, only the selected Codec
will be applied.
Packet Size
-The amount of data contained in a single packet.
The default value is 20 ms, which means the data packet will
contain 20 ms voice information.
Voice Active Detector -
This function can detect if the voice
on both sides is active or not. If not, the router will do
something to save the bandwidth for other using. Click On to
invoke this function; click off to close the function.
Default SIP Account
You can set SIP accounts (up to six groups) on SIP Account
page. Use the drop down list to choose one of the profile names
for the accounts as the default one for this phone setting.
Play dial tone only when account registered -
Check this box
to invoke the function.
In addition, you can press the
Advanced
button to configure tone settings, volume gain, MISC
and DTMF mode.
Advanced
setting is provided for fitting the telecommunication custom for
the local area of the router installed. Wrong tone settings might cause inconvenience for users.
To set the sound pattern of the phone set, simply choose a proper region to let the system find
out the preset tone settings and caller ID type automatically. Or you can adjust tone settings
manually if you choose User Defined. TOn1, TOff1, TOn2 and TOff2 mean the cadence of the
tone pattern. TOn1 and TOn2 represent sound-on; TOff1 and TOff2 represent the sound-off.

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